The present invention relates to a transmission method and apparatus of a voice packet, for example, and more particularly to a transmission method and apparatus which is suitably used for a voice packet transmission which switches a transmission method for a voice packet with a header, depending on whether the header is compressed or not.
Evolved UTRAN System
As a communication system today, systems using IP (Internet Protocol) and related protocols thereof are being constructed because of easy handling and cost, and there is a tendency that voice data is transmitted more and more using packets, not only in a cable system, but also in a radio space of a radio system.
When voice data is assembled into packets, voice encoding processing is performed on voice waveforms every 20 ms, for example. An RTP/UDP/IP (Real-Time Protocol/User Data Protocol/Internet Protocol) header is added to the encoded voice data payload, and transmitted as a voice packet. In a size of a voice packet per 20 ms, a ratio of the header is high, and in IPv6, the size of the IP header increases even more compared with IPv4.
FIG. 10 is a block diagram depicting an EUTRAN (Evolved UTRAN) system, under development by the standardization group 3GPP, is comprised of eNB (evolved-UTRAN Node B) 1a to 1n, which are base stations, access gateway aGW (evolved-UTRAN Access Gateway) 2a to 2b, which controls a group of base stations, and IASA (Inter Access System Anchor) 3, which is an anchor of the entire network. The base stations eNB 1a to 1n have functions similar to a conventional base station NB and radio network controller RNC, and aGW 2a and 2b transfer messages between the user terminals 4a and 4b and IASA 3. IASA 3, which has functions of a router, is connected to IMS (IP Multimedia System) 5, and to HSS (Home Subscriber Server) 6 for storing profiles of subscribers. A header portion of a voice packet is compressed in PDCP (Packet Data Convergence Protocol) function units in aGW (access Gate Way). A header is further added to the voice packet in RLC and MAC layers of the base stations, but the size of the voice packet becomes considerably small by RTP/UDP/IP header compression in PDCP function units.
Protocol Stack and Header Compression
FIG. 11 is a diagram depicting a protocol stack of U-plane Data in the user terminal (UE) 4, base station apparatus (eNB) 1 and aGW 2, which consists of four layers: a physical layer (PHY), MAC (Medium Access Control) layer, RLC (Radio Link Control) layer, and PDCP (Packet Data Convergence Protocol) layer. Data transmission and reception is executed in the PDCP layer between the user terminal (UE) 4 and aGW 2, and data is transmitted/received in the RLC layer between the user terminal 4 and base station 1. The major functions of each protocol are as follows.
(1) PDCP: In the PDCP layer, the transmission side compresses the header of the higher protocol, attaches a sequence number, and sends the data. The receive side checks the sequence number, whereby discard processing for redundant reception is executed. Retransmission is not performed in the PDCP layer.
FIG. 12 is a diagram depicting header compression, where (A) of FIG. 12 is a packet before header compression, in which IPv4 header (or IPv6 header) H1, UDP header H2 and RTP header H3 are attached to a voice payload PL, and (B) of FIG. 12 is a packet after header compression, in which the compressed header Hc is attached to the voice payload PL.
In the beginning, a packet, where headers H1 to H3 are attached to the voice data payload PL which is voice-encoded every 20 ms, is transmitted. The content of each header is divided into a portion to be unchanged and a portion to be changed. Therefore by attaching the entire content of the header to the voice payload PL and sending it only for a first time, and then attaching only the content to be changed to the voice payload, the header portion can be compressed. For example, the header portion size can be compressed down to about a 1 byte size in a case where only the RTP sequence number is sent all the time. However, if a transmission error occurs or if the content to be transmitted in a header is partially changed during transmitting voice payload with the compressed header Hc attached, the receive side cannot restore the header before compression using the compressed header. In such a case, the transmission side must send the uncompressed full size header. If restoring the header fails, the receive side sends feedback to the transmission side, and notifies the failure of header restoration in the receive side.
(2) RLC: The RLC layer has a layer having a retransmission function, and a new sequence number in the RLC layer is attached based on the sequence number attached to the data from PDCP, and this data is sent. Using this sequence number, the receive side notifies the transmission confirmation signal (Ack/Nack signal) to indicate normal reception/abnormal reception of data to the transmission side. The transmission side discards the data being held if the Ack signal is returned, or retransmits the data being held if the Nack signal is returned.
(3) MAC: The MAC layer is a layer to multiplex/demultiplex data in the RLC layer. In other words, the transmission side multiplexes data in the RLC layer to generate transmission data, and the receive side demultiplexes the receive data in the MAC layer into data in the RLC layer.
(4) PHY: The PHY layer is a layer to transmit/receive data via radio between the user terminal 4 and the base station 1, and converts the MAC layer data into radio data, or converts the radio data into MAC layer data.
Transmission Method and Subframe in Radio Blocks
In the downlink of the radio access portion of the Evolved UTRAN system, OFDM (Orthogonal Frequency Division Multiplex) is used. FIG. 13 is a diagram depicting a subframe in the downlink of the radio access portion of EUTRAN, in which the abscissa is the frequency (transmission band of the downlink), and the ordinate is the time, and five subframes are shown. Each subframe is comprised of a predetermined number of OFDM symbols, which are not illustrated. In the diagram, common pilots appear only in the front of every sub-frame. However, actually common pilots are included in common control signal region and individual data region as well.
An OFDM signal in the 20 MHz width radio transmission band (system transmission band) is sent by 1201 subcarriers. This 20 MHz width transmission band is divided into about 100 subbands (or Resource Blocks), and one or a plurality of subbands is/are used for data transmission to a terminal. It is assumed that one subband (or Resource Block) consists of 12 subcarriers. A subframe length is 1 ms, and a common pilot is sent through the entire system transmission band.
Each subframe SF includes a common pilot CPL, common control signal CCS, individual data addressed to each terminal UDT, individual control data UCT or voice data. The common pilot CPL is used for SIR measurement and synchronous demodulation at the receive side, and common control signal CCS is a control signal common to all terminals, and includes user data position information. The user data position information is information to notify the subcarrier or subband in which the user data is sent, to the terminal, and the terminal checks whether data addressed to this terminal exists, referring to this position information, and if it exists, the terminal receives the individual data/control signal UDT/UCT or voice signal addressed to this terminal from the specified subband.
OFDM Transmission/Receive System
FIG. 14 is a block diagram depicting a transmission apparatus in an OFDM communication system. A data modulation unit 11 modulates transmission data (user data or control data) based on QPSK/16QAM/64QAM data modulation, for example, and converts it into complex base band signals (symbols) having in-phase components and quadrature components. A time division multiplex unit 12 multiplexes pilot data of a plurality of symbols into data symbols using time and frequency division multiplexing. A serial to parallel conversion unit 13 converts the input data into M symbols of parallel data, and outputs M number of subcarrier samples. An IFFT (Inverse Fast Fourier transform) unit 14 performs IFFT (Inverse Fast Fourier Transform) processing on the subcarrier samples which are input in parallel, combines the results of the IFFT processing into a discrete time signal (referred to as an OFDM signal), and outputs it. A guard interval insertion unit 15 inserts a guard interval into the M symbols of the OFDM signal which is input from the IFFT unit, a transmission unit (TX) 16 converts the OFDM signal in which the guard interval is inserted from digital to analog, converts the frequency of the OFDM signal from the base band to a radio band, amplifies it, and sends it via a transmission antenna 17.
FIG. 15 is a block diagram of an OFDM receive apparatus. A signal which is output from a transmission antenna 7 is received by a receive antenna 18 of the receive apparatus via a fading channel (transmission line), and a receive circuit (Rx) 19 converts the RF signal received by the antenna into a base band signal, converts this base band signal from analog into digital, and outputs it. An FFT timing synchronous circuit 20 detects an FFT timing using a time domain signal which is output from the receive circuit 19, and a symbol generating unit 21 deletes GI, and generates an OFDM symbol at this FFT timing, and inputs it to the FFT unit 22. The FFT unit 22 performs FFT processing on each generated OFDM symbol, and converts it into subcarrier samples S0 to SM-1 in a frequency domain. A channel estimation circuit 23 calculates correlation of pilot symbols which are received with a predetermined interval and a known pilot pattern, whereby a channel estimation is performed for each subcarrier, and a channel compensation circuit 24 compensates the channel fluctuation of data symbols using the channel estimation value. By the above mentioned processing, transmission data distributed to each subcarrier is demodulated. Hereafter the demodulated subcarrier signal is converted into serial data, and is then decoded, although this is not illustrated.
Persistent Scheduling
In the radio systems of the future, it will be common to perform packet transmission in radio blocks based on scheduling. There are two types of data: RT (Real-Time) data in which the delay characteristic is most important; and NRT (Non-Real-Time) data in which throughput is most important.
In the case of a voice data packet which is transmitted with a predetermined interval, it is inefficient to send the data based on the scheduling that decides the transmission timing or the place (subband) in the frequency domain in packet units for each transmission, but it is efficient to send voice data packets at a predetermined period and place. This is because a control signal to indicate such information as the transmission timing and place, used for transmission of a voice packet, need not be sent for each voice packet.
It is also inefficient to send a voice data packet based on the scheduling that decides the modulation method and encoding rate in packet units. This is because the size of the voice packet data having a compressed header is small, so if the data is sent by changing the modulation method and encoding rate in packet units, the relative size of the control signal to indicate the modulation method and encoding rate used for the voice packet becomes too large with respect to the size of the voice packet.
This aspect is described using the example in FIG. 16. As (A) of FIG. 16 shows, a subframe has a control signal area 31 where scheduling information (timing, place in frequency domain, modulation method, encoding rate) is mapped, and a data area 32 where individual data and voice data are transmitted. The size of the control signal area 31 is limited, and if it is assumed that only a maximum N number of scheduling information S0 to SN-1 can be mapped, as shown in (B) of FIG. 16, then data (individual data and voice data) DT0 to DTN-1 addressed to only a maximum N number of terminals can be mapped in the data area 32. If it is non-real-time data, of which data size is large, the data area 32 becomes full before the number of data reaches N, and the data can be transmitted by fully using this data area. In the case of compressed voice packet data, of which data size is small, on the other hand, N or more number of voice packet data can be mapped in the data area 32. However since only a maximum N number of scheduling information S0, to SN-1, can be mapped in the control signal area 31, as (C) of FIG. 16 shows, voice packet data SD0 to SDN-1 addressed to only N number of terminals can be mapped in the data area 32, and space is generated in the data area 32, which makes data transmission inefficient.
A possible method is to map scheduling information in the control signal area 31 only when communication is first started, and include the scheduling information in the voice data thereafter, as shown in (D) of FIG. 16. According to this method, data can be transmitted fully utilizing the data area 32. However efficiency is not good, since the size of the scheduling information S0, to SN-1, is large.
Because of this, the following method has been proposed. When a voice packet is transmitted, the time axis is delimited with a predetermined time length, a transmission timing (period) of a voice packet PKT to be transmitted and a transmission place in the frequency domain are fixed in each time block T1, T2, T2, . . . , and a modulation method and encoding rate to be applied to a voice packet which is transmitted within the time block are decided and fixed, as shown in FIG. 17. And it is enabled to change the transmission timing, transmission place, demodulation method, encoding rate, transmission power or the like within each time block. Hence the information on the transmission timing, transmission place, modulation method, encoding rate and transmission power is specified in the time block by a control signal CS, which is attached to a voice packet PKT in the beginning of the time block, and sent.
In the case of using this transmission method, if a control signal to indicate the transmission method, modulation method or the like, which is transmitted in each time block, is received within the time block once, the receive side of the voice packet PKT can receive only the voice packet PKT thereafter within this time block, and transmission efficiency is improved. The standardization group 3GPP refers to this transmission method as a “voice packet transmission” based on persistent scheduling (Non-patent Document 1: 3GPP TR25.814). The control signal CS is attached to the voice packet PKT in the higher layer.
In FIG. 17, the control signal CS is attached to the first voice packet in a time block, but as FIG. 18 shows, the transmission timing, transmission place, modulation method, encoding rate and transmission power in the subsequent time block may be specified by the control signal CS attached to the last voice packet PKT in a time block.
FIG. 19 is a diagram describing a voice packet being transmitted in a predetermined period and in a same subcarrier (subband) in a time block. A voice packet PKT1 addressed to a terminal 1 is transmitted in a frequency band F1 in a time period T11, and a voice packet PKT2 addressed to a terminal 2 is transmitted in a frequency band F2 in a time period T22.
Problems
In EUTRAN, an RTP/UDP/IP header is compressed in PDCP function units in aGW, but a header attached at least to the first voice packet cannot be compressed. Then the RTP/UDP/IP header of each subsequent voice packet is compressed, but as described in connection with FIG. 12, a situation to require transmitting an uncompressed full sized header occurs non-periodically. Generally a size of a voice packet in which an uncompressed full sized header is attached is slightly less than double the size of a voice packet in which a compressed header is attached.
In the voice packet transmission method described in FIG. 17, if a situation to require transmitting a large sized voice packet in which a full sized header is attached suddenly occurs while a small sized voice packet of which header is compressed is being transmitted in a predetermined frequency band, transmission is disabled since a frequency band to transmit this large sized voice packet is not secured.
Unless this voice packet with a full sized header is transmitted, the header attached to the subsequent voice packet cannot be compressed. If a large frequency band is secured so that a voice packet with a fill sized header can be transmitted, on the other hand, unnecessary space must be secured when a small sized voice packet with a compressed header is transmitted, so a radio resource is wasted, and the significance of header compression is lost.